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Preamp low pass filter
I have just acquired a slightly beat-up Cambridge A75 preamp (thanks Rob!).
Aside from some tidying up, this will form part of my intended system upgrade to a bi-amped setup with a low-pass filter on the bass end. I'm attracted to fitting a second stereo output stage and filter to the preamp - that would let me try various amps for driving the bass speakers without having to deal with incorporating the filter into each. Although I can operate a soldering iron and a multimeter, I'm not terribly clued up on silicon electronics... I'm sure there are chips that can take an input from the PCB and drive a second pair of outputs to the same spec as the existing outputs - but what ones would be suitable? Those low noise mosfet 741-thingies? Is there a chip that I can use to construct a filter which doesn't introduce the sort of phase problems that a speaker-side crossover would? How much scope is there for being able to tailer the slope of the filter, and make the slope selectable? Can a subsonic shelf be incorporated? It would be good to be able to switch the filter off and have the second output provide full bandwidth. I'd be grateful for suggestions of other ways to approach the filtering, suitable chips, maybe some web sites that have circuit / schematics. -- Wally www.artbywally.com www.wally.myby.co.uk/music |
Preamp low pass filter
Wally wrote:
I have just acquired a slightly beat-up Cambridge A75 preamp (thanks Rob!). Aside from some tidying up, this will form part of my intended system upgrade to a bi-amped setup with a low-pass filter on the bass end. I'm attracted to fitting a second stereo output stage and filter to the preamp - that would let me try various amps for driving the bass speakers without having to deal with incorporating the filter into each. Although I can operate a soldering iron and a multimeter, I'm not terribly clued up on silicon electronics... I'm sure there are chips that can take an input from the PCB and drive a second pair of outputs to the same spec as the existing outputs - but what ones would be suitable? Those low noise mosfet 741-thingies? NE5532 would be my choice. Ian Is there a chip that I can use to construct a filter which doesn't introduce the sort of phase problems that a speaker-side crossover would? Very difficult to design a filter that changes amplitude with frequency without affecting phase. Laws of physics type stuff. How much scope is there for being able to tailer the slope of the filter, and make the slope selectable? No problem provided steps of about 6dB/octave are acceptable. Can a subsonic shelf be incorporated? No problem, in fact hard not to do this. It would be good to be able to switch the filter off and have the second output provide full bandwidth. Bypass is easy too. Ian |
Preamp low pass filter
Ian Bell wrote:
Very difficult to design a filter that changes amplitude with frequency without affecting phase. Laws of physics type stuff. Easy enough to do with a digital signal. A quick demo would be to low-pass a file with Goldwave or similar, reverse the file, filter it again and reverse it again. In the analogue world: You can keep the low-pass and high-pass outputs in phase with a 4th-order filter. I got a cheap Soundtech active crossover, which does this, on ebay. It's easy enough to build one though. There's some interesting stuff at Siegfried Linkwitz's site: http://www.linkwitzlab.com -- Roger. |
Preamp low pass filter
In message , Wally
writes I have just acquired a slightly beat-up Cambridge A75 preamp (thanks Rob!). Aside from some tidying up, this will form part of my intended system upgrade to a bi-amped setup with a low-pass filter on the bass end. I'm attracted to fitting a second stereo output stage and filter to the preamp - that would let me try various amps for driving the bass speakers without having to deal with incorporating the filter into each. Although I can operate a soldering iron and a multimeter, I'm not terribly clued up on silicon electronics... I'm sure there are chips that can take an input from the PCB and drive a second pair of outputs to the same spec as the existing outputs - but what ones would be suitable? Those low noise mosfet 741-thingies? Is there a chip that I can use to construct a filter which doesn't introduce the sort of phase problems that a speaker-side crossover would? How much scope is there for being able to tailer the slope of the filter, and make the slope selectable? Can a subsonic shelf be incorporated? It would be good to be able to switch the filter off and have the second output provide full bandwidth. I'd be grateful for suggestions of other ways to approach the filtering, suitable chips, maybe some web sites that have circuit / schematics. If you want an easy way to design and simulate low/high/band-pass filters of either Butterworth. Bessel or Chebychev with a conventional op-amp architecture with first to eighth order response, then go to the Microchip website and download 'Filterlab' (it's free). You fill in a table showing the -3dB points and the order and it does the rest. It even draws the schematic for you. -- Chris Morriss |
Preamp low pass filter
In article , Chris Morriss
writes In message , Wally writes I have just acquired a slightly beat-up Cambridge A75 preamp (thanks Rob!). Aside from some tidying up, this will form part of my intended system upgrade to a bi-amped setup with a low-pass filter on the bass end. I'm attracted to fitting a second stereo output stage and filter to the preamp - that would let me try various amps for driving the bass speakers without having to deal with incorporating the filter into each. Although I can operate a soldering iron and a multimeter, I'm not terribly clued up on silicon electronics... I'm sure there are chips that can take an input from the PCB and drive a second pair of outputs to the same spec as the existing outputs - but what ones would be suitable? Those low noise mosfet 741-thingies? Is there a chip that I can use to construct a filter which doesn't introduce the sort of phase problems that a speaker-side crossover would? How much scope is there for being able to tailer the slope of the filter, and make the slope selectable? Can a subsonic shelf be incorporated? It would be good to be able to switch the filter off and have the second output provide full bandwidth. I'd be grateful for suggestions of other ways to approach the filtering, suitable chips, maybe some web sites that have circuit / schematics. If you want an easy way to design and simulate low/high/band-pass filters of either Butterworth. Bessel or Chebychev with a conventional op-amp architecture with first to eighth order response, then go to the Microchip website and download 'Filterlab' (it's free). You fill in a table showing the -3dB points and the order and it does the rest. It even draws the schematic for you. If you're attempting to put more modern IC's where older ones once went make sure to run a scope over them to see if their not "hooting" in the MHz region which they seem to be very good at!.... -- Tony Sayer |
Preamp low pass filter
Plenty of "Pro Audio" crossover that are perfect for what you ask. Not
expensive, either. |
Preamp low pass filter
Ian Bell wrote:
I'm sure there are chips that can take an input from the PCB and drive a second pair of outputs to the same spec as the existing outputs - but what ones would be suitable? Those low noise mosfet 741-thingies? NE5532 would be my choice. Cheers. Very difficult to design a filter that changes amplitude with frequency without affecting phase. Laws of physics type stuff. Oh well - I was hoping there might be some clever chippery that could do that sort of thing. How much scope is there for being able to tailer the slope of the filter, and make the slope selectable? No problem provided steps of about 6dB/octave are acceptable. I dare say - I want to be able to match the slope to whatever response I get out of the upper bass / mid speakers. Can a subsonic shelf be incorporated? No problem, in fact hard not to do this. Bypass is easy too. Good-o. -- Wally www.artbywally.com www.wally.myby.co.uk/music |
Preamp low pass filter
Old Fart at Play wrote:
In the analogue world: You can keep the low-pass and high-pass outputs in phase with a 4th-order filter. I got a cheap Soundtech active crossover, which does this, on ebay. It's easy enough to build one though. I'll have a look for ready-made active crossovers and see how much the cost. There's some interesting stuff at Siegfried Linkwitz's site: http://www.linkwitzlab.com That looks like a good read... -- Wally www.artbywally.com www.wally.myby.co.uk/music |
Preamp low pass filter
Chris Morriss wrote:
If you want an easy way to design and simulate low/high/band-pass filters of either Butterworth. Bessel or Chebychev with a conventional op-amp architecture with first to eighth order response, then go to the Microchip website and download 'Filterlab' (it's free). Cheers for that - downloaded. -- Wally www.artbywally.com www.wally.myby.co.uk/music |
Preamp low pass filter
Old Fart at Play wrote:
Ian Bell wrote: Very difficult to design a filter that changes amplitude with frequency without affecting phase. Laws of physics type stuff. Easy enough to do with a digital signal. A quick demo would be to low-pass a file with Goldwave or similar, reverse the file, filter it again and reverse it again. In the analogue world: You can keep the low-pass and high-pass outputs in phase with a 4th-order filter. I got a cheap Soundtech active crossover, which does this, on ebay. It's easy enough to build one though. There's some interesting stuff at Siegfried Linkwitz's site: http://www.linkwitzlab.com Which include some very nice graphs showing both the frequency and *phase* response. Ian |
Preamp low pass filter
In message , Old Fart at Play
writes You can keep the low-pass and high-pass outputs in phase with a 4th-order filter. Not without an additional all-pass filter you can't. And with an added all-pass you can make a second order crossover have no phase difference between the LP and HP sections if you want. -- Chris Morriss |
Preamp low pass filter
In article , Chris Morriss
wrote: In message , Old Fart at Play writes You can keep the low-pass and high-pass outputs in phase with a 4th-order filter. Not without an additional all-pass filter you can't. And with an added all-pass you can make a second order crossover have no phase difference between the LP and HP sections if you want. However you may not actually want that. :-) Personally, for active filtering, I'd tend to prefer using a LPF, then creating a HPF output by subtracting the LPF output from the input. The result if you keep the levels matched is a LP and HP pair of signals whose vector sum always equals the input. Thus the combined result shows no phase errors due to the filtering. For the actual filters I tend to lift the basic designs from the Active Filter Cookbook by Don Lancaster. Slainte, Jim -- Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm Audio Misc http://www.st-and.demon.co.uk/AudioMisc/index.html Armstrong Audio http://www.st-and.demon.co.uk/Audio/armstrong.html Barbirolli Soc. http://www.st-and.demon.co.uk/JBSoc/JBSoc.html |
Preamp low pass filter
Chris Morriss wrote:
In message , Old Fart at Play writes You can keep the low-pass and high-pass outputs in phase with a 4th-order filter. Not without an additional all-pass filter you can't. And with an added all-pass you can make a second order crossover have no phase difference between the LP and HP sections if you want. Perhaps you would like to refer to the Loudspeaker Design Cookbook which has graphs of amplitude and phase for various filters including the fourth order Linkwitz-Riley filter. -- Roger. |
Preamp low pass filter
Old Fart at Play wrote:
Chris Morriss wrote: In message , Old Fart at Play writes You can keep the low-pass and high-pass outputs in phase with a 4th-order filter. Not without an additional all-pass filter you can't. And with an added all-pass you can make a second order crossover have no phase difference between the LP and HP sections if you want. Perhaps you would like to refer to the Loudspeaker Design Cookbook which has graphs of amplitude and phase for various filters including the fourth order Linkwitz-Riley filter. Remember that it's the acoustical 4th order Linkwitz-Riley that has the so-called zero phase difference, not some speaker implemented with 4th order crossovers. The acoustical 4th order is implemented with second-order crossovers. The other second-order filters are the roll-offs of the speakers themselves. |
Preamp low pass filter
In message , Old Fart at Play
writes Chris Morriss wrote: In message , Old Fart at Play writes You can keep the low-pass and high-pass outputs in phase with a 4th-order filter. Not without an additional all-pass filter you can't. And with an added all-pass you can make a second order crossover have no phase difference between the LP and HP sections if you want. Perhaps you would like to refer to the Loudspeaker Design Cookbook which has graphs of amplitude and phase for various filters including the fourth order Linkwitz-Riley filter. OK, I've looked at that, and it doesn't support what you say at all. I think you are getting confused between a crossover that keeps the outputs of all the sections in phase at all frequencies (which can only be done with even-ordered networks, and then only in conjunction with all-pass phase-correction networks) and the family of crossovers that attempt to sum to a flat frequency and phase response, even though the individual outputs have phase differences between them. (By the way, you can't get a passive filter to have LP and HP outputs in phase with each other at all frequencies, as it's not possible to produce the right sort of all-pass network with passive components.) You can of course make passive networks that have a summed flat magnitude and phase response, but this is a different thing entirely. Although all-pass phase corrected active crossovers can be made, they are not universally liked, as the extra group-delay added by the phase-compensating all-pass networks mean that the total variation in phase across the whole audio band can be very considerable. (Whether or not this is audible on music is debatable). -- Chris Morriss |
Preamp low pass filter
Jim Lesurf wrote:
Personally, for active filtering, I'd tend to prefer using a LPF, then creating a HPF output by subtracting the LPF output from the input. The result if you keep the levels matched is a LP and HP pair of signals whose vector sum always equals the input. Thus the combined result shows no phase errors due to the filtering. That's a neat trick. (Maybe tri-amping isn't such a bad idea...) For the actual filters I tend to lift the basic designs from the Active Filter Cookbook by Don Lancaster. taking notes -- Wally www.artbywally.com www.wally.myby.co.uk/music |
Preamp low pass filter
In message , Jim Lesurf
writes However you may not actually want that. :-) Personally, for active filtering, I'd tend to prefer using a LPF, then creating a HPF output by subtracting the LPF output from the input. The result if you keep the levels matched is a LP and HP pair of signals whose vector sum always equals the input. Thus the combined result shows no phase errors due to the filtering. For the actual filters I tend to lift the basic designs from the Active Oh yes, I quite agree, a complex phase-compensated crossover has only one advantage: it does help keep down vertical lobing problems. As Arnie has also said, it does also depend on the inherent amplitude and phase response of the drivers. I use constant-voltage subtraction crossovers, but without any phase compensation they do force one of the outputs to only roll off at 6db per octave. -- Chris Morriss |
Preamp low pass filter
Chris Morriss wrote:
In message , Old Fart at Play writes Chris Morriss wrote: In message , Old Fart at Play writes You can keep the low-pass and high-pass outputs in phase with a 4th-order filter. Not without an additional all-pass filter you can't. And with an added all-pass you can make a second order crossover have no phase difference between the LP and HP sections if you want. Perhaps you would like to refer to the Loudspeaker Design Cookbook which has graphs of amplitude and phase for various filters including the fourth order Linkwitz-Riley filter. OK, I've looked at that, and it doesn't support what you say at all. I think you are getting confused between a crossover that keeps the outputs of all the sections in phase at all frequencies (which can only be done with even-ordered networks, and then only in conjunction with all-pass phase-correction networks) and the family of crossovers that attempt to sum to a flat frequency and phase response, even though the individual outputs have phase differences between them. Have the laws of physics changed since my LDC4 was published? Section 7.21:Combined response of two-way crossovers "....exhibit a high-pass and low-pass phase relationship which is in-phase." Graphs 7.58 and 7.59 show what I mean. -- Roger. |
Preamp low pass filter
In message , Old Fart at Play
writes Chris Morriss wrote: In message , Old Fart at Play writes Chris Morriss wrote: In message , Old Fart at Play writes You can keep the low-pass and high-pass outputs in phase with a 4th-order filter. Not without an additional all-pass filter you can't. And with an added all-pass you can make a second order crossover have no phase difference between the LP and HP sections if you want. Perhaps you would like to refer to the Loudspeaker Design Cookbook which has graphs of amplitude and phase for various filters including the fourth order Linkwitz-Riley filter. OK, I've looked at that, and it doesn't support what you say at all. I think you are getting confused between a crossover that keeps the outputs of all the sections in phase at all frequencies (which can only be done with even-ordered networks, and then only in conjunction with all-pass phase-correction networks) and the family of crossovers that attempt to sum to a flat frequency and phase response, even though the individual outputs have phase differences between them. Have the laws of physics changed since my LDC4 was published? Section 7.21:Combined response of two-way crossovers "....exhibit a high-pass and low-pass phase relationship which is in-phase." Graphs 7.58 and 7.59 show what I mean. Ok, I can see where your confusion is coming from. No, the laws of physics haven't changed, but the LS cookbook doesn't make things clear. If you read on in the same section, you'll see that it says "the two sections sum together flat when the level of both filters is down 6dB at the crossover frequency". This is the crux of the issue. To get a flat amplitude response from an even order filter, the crossover frequency should be at the -6dB point, but to get the two outputs to be in-phase (actually 180 out of phase, but this is cured by turning the connections round on either the tweeter or the bass unit), the crossover frequency needs to be at the -3dB point. Here's an example. It's for a second order Butterworth. (And remember that a 4th order L_R is simply two identical 2nd order Butterworths in series) If the crossover is at the -3dB point, the phase is at 90 degrees at that point, and the HP and LP will be consistently 180 degrees out of phase, BUT the magnitude will sum to have a 3dB hump. If the crossover is at the 6dB point, the magnitude will sum to be flat, but as the phase shift at the -6dB point is 110 degrees (rather than 90) the phase shifts of the two outputs will not track. In reality a passive crossover is tweaked to give a compromise (and to allow for the amplitude/phase characteristics of the drivers...if you've got a competent design team that is. An active crossover can be made to have perfect phase tracking between the HP and LP outputs by judicious use of all-pass networks. (Though as Jim says, that may not be what you want for best fidelity) -- Chris Morriss |
Preamp low pass filter
Chris Morriss wrote:
In message , Old Fart at Play writes Chris Morriss wrote: In message , Old Fart at Play writes Chris Morriss wrote: In message , Old Fart at Play writes You can keep the low-pass and high-pass outputs in phase with a 4th-order filter. Not without an additional all-pass filter you can't. And with an added all-pass you can make a second order crossover have no phase difference between the LP and HP sections if you want. Perhaps you would like to refer to the Loudspeaker Design Cookbook which has graphs of amplitude and phase for various filters including the fourth order Linkwitz-Riley filter. OK, I've looked at that, and it doesn't support what you say at all. I think you are getting confused between a crossover that keeps the outputs of all the sections in phase at all frequencies (which can only be done with even-ordered networks, and then only in conjunction with all-pass phase-correction networks) and the family of crossovers that attempt to sum to a flat frequency and phase response, even though the individual outputs have phase differences between them. Have the laws of physics changed since my LDC4 was published? Section 7.21:Combined response of two-way crossovers "....exhibit a high-pass and low-pass phase relationship which is in-phase." Graphs 7.58 and 7.59 show what I mean. Ok, I can see where your confusion is coming from. No, the laws of physics haven't changed, but the LS cookbook doesn't make things clear. If you read on in the same section, you'll see that it says "the two sections sum together flat when the level of both filters is down 6dB at the crossover frequency". This is the crux of the issue. To get a flat amplitude response from an even order filter, the crossover frequency should be at the -6dB point, but to get the two outputs to be in-phase (actually 180 out of phase, but this is cured by turning the connections round on either the tweeter or the bass unit), the crossover frequency needs to be at the -3dB point. Here's an example. It's for a second order Butterworth. (And remember that a 4th order L_R is simply two identical 2nd order Butterworths in series) If the crossover is at the -3dB point, the phase is at 90 degrees at that point, and the HP and LP will be consistently 180 degrees out of phase, BUT the magnitude will sum to have a 3dB hump. If the crossover is at the 6dB point, the magnitude will sum to be flat, but as the phase shift at the -6dB point is 110 degrees (rather than 90) the phase shifts of the two outputs will not track. Chris, you are still confused but nearly there. As you say, a 4th order L-R is two 2nd order Butterworths. At the crossover frequency a B2 filter is -3dB and 90 degrees phase shift. Therefore the 4LR is -6dB and 180 degrees. The LP and HP outputs are in phase at all frequencies and the voltage sum is constant. HTH, Roger. |
Preamp low pass filter
In message , Old Fart at Play
writes Chris, you are still confused but nearly there. As you say, a 4th order L-R is two 2nd order Butterworths. At the crossover frequency a B2 filter is -3dB and 90 degrees phase shift. Therefore the 4LR is -6dB and 180 degrees. The LP and HP outputs are in phase at all frequencies and the voltage sum is constant. HTH, Roger. You are quite correct, I'll go and hang my head in shame now :-( For a 4-th order L-R, you do indeed cross over at the -6dB point and the outputs of the LP and HP sections are always in phase. (And the magnitude sums to be flat). (I even went and SPICEd it to check!) -- Chris Morriss |
Preamp low pass filter
In article , Chris Morriss
wrote: In message , Jim Lesurf writes However you may not actually want that. :-) Personally, for active filtering, I'd tend to prefer using a LPF, then creating a HPF output by subtracting the LPF output from the input. The result if you keep the levels matched is a LP and HP pair of signals whose vector sum always equals the input. Thus the combined result shows no phase errors due to the filtering. For the actual filters I tend to lift the basic designs from the Active Oh yes, I quite agree, a complex phase-compensated crossover has only one advantage: it does help keep down vertical lobing problems. I would put this slightly differently. The 'lobing' problem arises as a result of having an 'array' of speakers in operation in the crossover frequency region. There will always tend to be a frequency region where the two units are radiating similar powers. If the speakers are not very close (in wavelength terms) lobing is then inevitable. The phasing in this region won't prevent lobing, it will just displace the maxima and minima in angular terms w.r.t. the line through the speakers and the speaker plane. As Arnie has also said, it does also depend on the inherent amplitude and phase response of the drivers. The key point here for me is the phase responses of the two drivers in the frequency region where they are tending to radiate similar powers. If they are 'in phase' at this point, then ensuring the vector sum is unchanged should mean that the 'far field' power sent normal to the line through the speaker units (i.e. towards the nominal listener) will be correct. However the above makes assumptions about what is the case. So, for example, if the speakers have phase delays that differ when they are radiating similar amounts, you'd need to change what you are giving them. We also have to worry about where the listener may be and the room acoustic. All of this is another reason why I'm not really a fan of 'dynamic' speakers. :-) The advantage of the method I prefer is that it ensures both constant amplitude sum (for the correct unit phase behaviour) and constant total power. Does this by ensuring the vector sum gain from the filtering is frequency independent. However this may not be what a specific speaker requires. Above said, for electronic crossovers, I'd tend to do it this way, then add any required modifiers to 'pre-correct' the split signals before delivering them to the power amps and units... I use constant-voltage subtraction crossovers, but without any phase compensation they do force one of the outputs to only roll off at 6db per octave. The advantage of higher orders is they can cut down to size of the region where we have an (unwanted) array effect. However you can do this using my approach, and it saves money as you only need one high-order LPF and then get the HPF that matches it 'for free'. :-) Slainte, Jim -- Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm Audio Misc http://www.st-and.demon.co.uk/AudioMisc/index.html Armstrong Audio http://www.st-and.demon.co.uk/Audio/armstrong.html Barbirolli Soc. http://www.st-and.demon.co.uk/JBSoc/JBSoc.html |
Preamp low pass filter
In message , Jim Lesurf
writes The advantage of higher orders is they can cut down to size of the region where we have an (unwanted) array effect. However you can do this using my approach, and it saves money as you only need one high-order LPF and then get the HPF that matches it 'for free'. :-) Slainte, Jim But even if your HPF (say) is a 4-th order, the LPF you get by subtraction is still only a first order. The phase shift of the summed output is zero, and it does sum flat of course, so it still is a good thing. (It took me ages to work this out, but it is correct, and a quick SPICE simulation shows it) As I said in an earlier post, I do this on a homemade two-way (M-T-M) that I use. A second order HPF for the tweeter, and the subtractive LPF for the Bass units. I put the main filter as the HPF to roll-off the tweeter reasonably quickly, whereas I wasn't too worried about the low-order roll-of of the subtractive LP output. -- Chris Morriss |
Preamp low pass filter
Chris Morriss wrote:
In message , Old Fart at Play writes Chris, you are still confused but nearly there. As you say, a 4th order L-R is two 2nd order Butterworths. At the crossover frequency a B2 filter is -3dB and 90 degrees phase shift. Therefore the 4LR is -6dB and 180 degrees. The LP and HP outputs are in phase at all frequencies and the voltage sum is constant. HTH, Roger. You are quite correct, I'll go and hang my head in shame now :-( For a 4-th order L-R, you do indeed cross over at the -6dB point and the outputs of the LP and HP sections are always in phase. (And the magnitude sums to be flat). Thanks. Of course it only works if the woofer and tweeter work well for an octave or so beyond the crossover frequency, otherwise as Arny said, you have to consider the acoustical output. -- Roger. |
Preamp low pass filter
In article , Chris Morriss
wrote: This is the crux of the issue. To get a flat amplitude response from an even order filter, the crossover frequency should be at the -6dB point, Not quite. The only general requirement would be that the vector sum adds up to the 0dB level. If you allow the components to be in quadrature, for example, this can occur if they cross at -3dB. A problem here is that there is often an ambiguity in discussions due to the coherent effects of the way the radiation pattern is affected by phase, and the mean (space-averaged) power around a room, and on axis. Ideally, you'd want to know the directional and amplitude/phase properties of the speakers, and the acoustics of the room, and the choice of listening location. Lacking these, you end up with having to choose a set of assumptions. Slainte, Jim -- Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm Audio Misc http://www.st-and.demon.co.uk/AudioMisc/index.html Armstrong Audio http://www.st-and.demon.co.uk/Audio/armstrong.html Barbirolli Soc. http://www.st-and.demon.co.uk/JBSoc/JBSoc.html |
Preamp low pass filter
In article , Chris Morriss
wrote: In message , Jim Lesurf writes The advantage of higher orders is they can cut down to size of the region where we have an (unwanted) array effect. However you can do this using my approach, and it saves money as you only need one high-order LPF and then get the HPF that matches it 'for free'. :-) Slainte, Jim But even if your HPF (say) is a 4-th order, the LPF you get by subtraction is still only a first order. I've not done this for a while, but that strikes me as rather odd (apology for the pun! :-) ) as a general claim. I think you may find it depends upon the details of the filter shape of the LPF filter, not just the order. IIRC when I did some filtering like this a while ago for analysis of the effects of HF on tweeters the HP anf LP sections (done this way) were of the same sort or roll-off slopes. May be mis-remembering, though... The phase shift of the summed output is zero, and it does sum flat of course, so it still is a good thing. (It took me ages to work this out, but it is correct, and a quick SPICE simulation shows it) I think this may depend upon some specific assumptions you may have made. However I'll be interested to hear what you can report on this. Consider designing a LPF that is approaching a 'brick wall' with a flat top near 0dB and a fall-off to, say, -60dB that occurs over a narrow range. Where the LPF is near 0dB the output from the 'HPF' must be very small. However as with transit the turnover region it rises to near 0dB. The narrower the transition region, the steeper the slope of the turnover of both the HP and LP outputs. OTOH if you choose a high order filter that has an inband 'droop' whose size scales up significantly with the order, you will, indeed reduce the slope of the HP output in the 'droop' region. But what matters here is the filter shape, not just the order. I've forgotten the latin for 'taking an absurd example' for the sake of extreme illustration. However I won't let my lack of decent classical education deter me from the following... :-) Imagine building an analogue version of one of the 96th order low pass filters used for digital. These can have an inband ripple that is very close to 0dB, yet die the death in the space of about 2kHz. If you were to apply the above methods to get the HP difference I doubt it would show just a first order rolloff as you went down into the low frequency range. I think the response would change very rapidly over the same 2kHz-ish band. Or am I missing something? Slainte, Jim -- Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm Audio Misc http://www.st-and.demon.co.uk/AudioMisc/index.html Armstrong Audio http://www.st-and.demon.co.uk/Audio/armstrong.html Barbirolli Soc. http://www.st-and.demon.co.uk/JBSoc/JBSoc.html |
Preamp low pass filter
In message , Jim Lesurf
writes In article , Chris Morriss wrote: In message , Jim Lesurf writes The advantage of higher orders is they can cut down to size of the region where we have an (unwanted) array effect. However you can do this using my approach, and it saves money as you only need one high-order LPF and then get the HPF that matches it 'for free'. :-) Slainte, Jim But even if your HPF (say) is a 4-th order, the LPF you get by subtraction is still only a first order. I've not done this for a while, but that strikes me as rather odd (apology for the pun! :-) ) as a general claim. I think you may find it depends upon the details of the filter shape of the LPF filter, not just the order. IIRC when I did some filtering like this a while ago for analysis of the effects of HF on tweeters the HP anf LP sections (done this way) were of the same sort or roll-off slopes. May be mis-remembering, though... The phase shift of the summed output is zero, and it does sum flat of course, so it still is a good thing. (It took me ages to work this out, but it is correct, and a quick SPICE simulation shows it) I think this may depend upon some specific assumptions you may have made. However I'll be interested to hear what you can report on this. It struck me as rather odd as well when it was first put to me. Do you have SPICE available to you? It's easy enough to make an arbitary LP or HP filter and use an "ideal subtractor" element to produce the other output. It's easy enough to verify it. I'll try it with a high order filter, (6-th order or so) and run the simulation. I guess your email address is valid, so I'll email you the gain/phase plot of the sim run. Regards, -- Chris Morriss |
Preamp low pass filter
In message , Jim Lesurf
writes I think this may depend upon some specific assumptions you may have made. However I'll be interested to hear what you can report on this. I've sent you an email with the SPICE results. They are in .wmf format. The email is 216k long so I hope you allow emails that size through! -- Chris Morriss |
Preamp low pass filter
In article , Chris Morriss
wrote: In message , Jim Lesurf writes I think this may depend upon some specific assumptions you may have made. However I'll be interested to hear what you can report on this. I've sent you an email with the SPICE results. They are in .wmf format. The email is 216k long so I hope you allow emails that size through! The size is no problem. The snag is that I don't use 'doze, so now will have to translate the WMFs and hope the translations comes out OK. ;-) Don't send me a different format (yet!), though. I'll have a bash when I can and reply in detail by email. Slainte, Jim -- Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm Audio Misc http://www.st-and.demon.co.uk/AudioMisc/index.html Armstrong Audio http://www.st-and.demon.co.uk/Audio/armstrong.html Barbirolli Soc. http://www.st-and.demon.co.uk/JBSoc/JBSoc.html |
Preamp low pass filter
In article , Chris Morriss
wrote: In message , Jim Lesurf writes I think this may depend upon some specific assumptions you may have made. However I'll be interested to hear what you can report on this. It struck me as rather odd as well when it was first put to me. Do you have SPICE available to you? I don't use SPICE much. I tend to write my own programs, or analyse things like filters using the response equations given in the books like the cookbook I mentioned. I don't think I have a copy of SPICE to hand that is 32-bit 'clean' for the OS/CPU I use. I'll investigate when I get a chance, though. I'll try it with a high order filter, (6-th order or so) and run the simulation. I guess your email address is valid, so I'll email you the gain/phase plot of the sim run. The files have arrived OK. :-) Slainte, Jim -- Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm Audio Misc http://www.st-and.demon.co.uk/AudioMisc/index.html Armstrong Audio http://www.st-and.demon.co.uk/Audio/armstrong.html Barbirolli Soc. http://www.st-and.demon.co.uk/JBSoc/JBSoc.html |
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